[eng]: does it forward PSTN-to-VoIP without going hook-off?

hello,

sorry for writing english in here

i would like to use Asterisk as the base call triggering device and let ANA (SPA-3000) work only as a gateway for incoming from PSTN calls.

does anyone know if this possible to configure ANA so that it does not create the PSTN call leg (does not answer to incoming PSTN call) but sends INVITEs immediately?

more precise:

if configured so (as "hotline"), SPA-3000 does the following:

1.answers the call on PSTN (makes off-hook)
2.dials in the assigned for forwarding VoIP address (sends INVITE message)

step #1 creates PSTN call leg for the caller (what is not right in
case if VoIP address not answering, then caller will be billed for
this), is there a way (for such "hotline") to avoid picking up the
call (step#1) completely (so if the PSTN line ringing, then only
step#2 performed; only in case if the VoIP address is answering the
call, then SPA goes off-hook on PSTN line)?

thanks,

tommmy.leee

thanks for the answering. actually googling does its job:

http://www.voip-info.org/tiki-print.php?page=Sipura+3000

but it is just a workaround... i think everybody who owns this device would be happy to have such a feature just in the firmware.

look at my mailng with sipura support guys below, maybe if somebody else writes on them willing the same then they turn out to implement it with next release of its firmware...


TOM to SIPURA SUPPORT

hello again,

what a pity. nevertheless, thank you very much for all your time you
spend answering me.

by the way, this feature is desired by many others, not only me.
people found a work around to do so BUT it does not work in 100%
cases, so maybe there is a possibility to implement it in the
firmware...

take a look here
http://www.voip-info.org/tiki-print.php?page=Sipura+3000

would be nice if you take this one as the wish from your faithful customers.

thank you so much again!

with best regards,
tommmy leee
- Hide quoted text -


On Tue, 08 Feb 2005 11:42:06 -0800, Sipura Support <[email protected]> wrote:
> no.
> The PSTN will always pickup prior to dialing out via VoIP.
> This behavior is not configureable.
>
> -Sipura Support
>
> tommmy leee wrote:
>
> > hello,
> >
> > thank you very much for your very prompt answer!
> >
> > actually i did this configuration but i meant another thing in my
> > previous mail (maybe my English is not good enough).
> >
> > here i try to formulate my question more precise.
> >
> > if configured so (as "hotline"), SPA-3000 does the following:
> >
> > 1.answers the call on PSTN (makes off-hook)
> > 2.dials in the assigned for forwarding VoIP address (sends INVITE message)
> >
> > step #1 creates PSTN call leg for the caller (what is not right in
> > case if VoIP address not answering, then caller will be billed for
> > this), is there a way (for such "hotline") to avoid picking up the
> > call (step#1) completely (so if the PSTN line ringing, then only
> > step#2 performed; only in case if the VoIP address is answering the
> > call, then SPA goes off-hook on PSTN line)?
> >
> > thank you very much in advance!!!
> >
> > best regards,
> > tommmy leee
> >
> >
> > On Mon, 07 Feb 2005 17:59:53 -0800, Sipura Support <[email protected]> wrote:
> >
> >>If your PSTN line is configured to have VoIP service,
> >>then you can do a call forward via this method described below:
> >>http://www.sipura.com/support/spa3000faq/Section_2.html#17
> >>
> >>-Sipura Support
> >>
> >>tommmy leee wrote:
> >>
> >>
> >>>dear Sirs,
> >>>
> >>>i am the happy customer of Sipura product (SPA-3000, fiirmware 2.0.10)
> >>>since a week but i was dissapointed to find that i can not configure
> >>>this device a way that it does not create FXO call leg in case when i
> >>>want to forward _all_ incoming from FXO (PSTN) calls to internal
> >>>SIP-phone (currently it is possible to forward a call from PSTN based
> >>>on caller ID but it is not a case i talk about).
> >>>
> >>>would be nice to have a "hot line" option in "PSTN-To-VoIP Gateway
> >>>Setup" section on the SPA-3000, when selected, then PSTN interface
> >>>would become dumb on incoming calls and send an invite request
> >>>immediately after PSTN caller ID is received (even caller ID is
> >>>suppressed), and only seize the PSTN line when the VoIP call to
> >>>internal SIP-phone is actually answered.
> >>>
> >>>thank you very much for your time and hope this wish will be regarded.
> >>>
> >>>nevertheless, just to repeat one more time, the product is great!
> >>>
> >>>with very best regards from Germany,
> >>>tommmy leee